Chapter 12 Transport protocols. Outline 12.1 introduction 12.2 TCP/IP protocol suite.

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Presentation transcript:

Chapter 12 Transport protocols

Outline 12.1 introduction 12.2 TCP/IP protocol suite

12.1 introduction Transmission control protocol (TCP) User datagram protocol (UDP) Real-time transport protocol(RTP) Real-time transport control protocol(RTCP)

12.2 TCP/IP protocol suite Protocol field: identifying the protocol to which the contents of the datagram relate Port numbers: identifying the application protocol to which the PDU contents relate Client port numbers are called ephemeral ports Server port numbers are known as well-known port numbers in the range 1 through 1023

12.3 TCP Reliable stream service:Each byte in the stream flowing in each direction is free of transmission errors and in the same sequence Each TCP entity divides the stream of bytes into blocks known as segments Maximum segment size(MSS) The default MSS is 536 bytes The TCP protocol includes a flow control procedure to ensure no data is lost

user services Socket :Each of the two peer user application protocols first creates a communications channel Figure 12.3 Once a socket has been created a socket descriptor is returned to the AP for the subsequent primitive calls Bind() has an address parameter(socket address) Listen() results in the local TCP entity creating a queue to hold incoming connection requests

user services Accept() is used to put the AP in the blocked state to be received from a client TCP entity The sequence of four primitives is a passive-open In order for the two TCP entities to relate each received segment to the correct connection, both TCP entities create a connection record for it Send() primitive is used to transfer a block of data to the send buffer When the server AP has finished sending data, it issues a close() primitive to release the other side of connection

protocol operation TCP protocol involves three operations Setting up a logical connection between two sockets Transferring blocks of data over this connection Closing down the logical connection Figure 12.4 Pseudo header: for an additional level of checking, some fields from the IP header are also included in the computation of the TCP checksum Pad byte of zero is added to the data field whenever the number of bytes in the original data field is odd

protocol operation Urgent data: when the URG flag is set in the code field, the number of bytes in the data field that follow the current sequence number Three-way handshake It sends a segment to the TCP in the server with SYN code bit on, the ACK bit off, and the chosen ISN(X) in the sequence field If the server AP is in the LISTEN state, the server TCP makes an entry of the ISN On receipt of the segment the client TCP enters the ESTABLISHED state

protocol operation On receipt of the ACK, the server TCP enters the ESTABLISHED state and both sides are ready to exchange data segments Simultaneous open: two Aps may try to establish a connection at the same time Window size advertisement: to inform the other TCP entity of the maximum number of bytes Figure 12.5

protocol operation Delayed acknowledgements:in order to reduce the number of segments that are send, a receiving TCP entity does not return an ACK segment immediately it receives an segment Nagle algorithm: in interactive applications, a number of characters that have been typed by the user waiting in the send buffer are transmitted in a single segment Figure 12.6

protocol operation It only retransmits a segment if it receives three duplicate ACKs for the same segment Send sequence variable V(S) is the sequence number field of the next new segment it sends Retransmission list holds segments waiting to be acknowledged V(R) indicates the sequence number it expects Receive list hold segments that are received out of sequence

protocol operation Figure 12.7 Fast retransmit: the retransmission occurs before the timer expires Retransmission timeout(RTO) interval: RTO is set at a value slightly greater than the interval between sending a packet and receiving an ACK The choice of RTO must be dynamic Exponential backoff algorithm

protocol operation Window size is determined by the amount of free space that is present in the receive buffer being used by the receiving TCP Flow control scheme ensures that there is always the required amount of free space in the receive buffer before the source sends the data Figure 12.8

protocol operation Reason for lost packets(congestion):With heavy traffic it temporary runs out of buffer storage for packets in the output queue associated with a line Congestion window: uses the rate of arrival of the ACKs relating to a connection to regulate the rate of entry of data segments Figure The sending TCP starts the transfer phase of a connection by sending a single segment

protocol operation If the ACK is received before the timer expires, W c is increased to two segments This phase is called slow start Slow start threshold(SST) is set to 64k bytes Assuming the SST is reached, this is taken as an indication that the path is not congested Congestion avoidance: It enters a second phase during which it increases by 1/W c segments for each ACK received

protocol operation Fast recovery: on receipt of the third duplicate ACK, the current W c value is halves When a retransmission timeout occurs, it is immediately reset to 1 segment Figure 12.11

protocol operation The AP which issues the first close() performs an called active close and the other is passive close Figure In fig12.12(b), it reduces the standard closure to a 3-way segment exchange rather than 4-way The disadvantage is that data may be lost at the passive side if both sides are closed On receipt of the related ACK segment, both sides enter the TIMED_WAIT state to wait for the 2MSL timer to expire

protocol operation Half-close: the local TCP initiates the closure of its side of the connection but leaves the other side in the ESTABLISHED state Persist timer Whenever the sending TCP sets its send window,Ws to zero, it starts a timer If a segment containing a window update is not received before timer expires, the sending TCP sends a window probe segment Figure 12.13

protocol operation Keepalive timer If the client host is switched off the connection from the server to the client will remain The default value of the keepalive timer is two hours The TCP in the server sends a probe segment to the client and sets the timer this time to 75 s This procedure is repeated and if no reply is received after 10 consecutive probes, the server terminates Figure 12.14

Figure Keepalive timer: application and operation.

protocol operation Silly window syndrome(SWD) in interactive applications a very small number of bytes being sent in each segment A receiving TCP is prevented from sending a window update until there is sufficient space in its buffer Figure Window scale option An option has been defined that enables a scaling factor to be applied to the value specified in the window size fields Figure 12.16

protocol operation Time-stamp option It is used with these implementations when a large window size is detected Figure SACK-permitted option With connections that have a large RTT associated with them, the delays involved each time a packet is lost or corrupted can be large Protection against wrapped sequence numbers(PAWS) A segment that is lost during one pass through the sequence numbers may be retransmitted during a later pass through the numbers

12.4 UDP There are no error or flow control procedures and no connection set up is required The maximum theoretical size of a UDP datagram is bytes, the maximum value supported by most implementations is 8192 bytes or less

12.5 RTP and RTCP RTP In a multicast call, each participants is called a contributing source(CSRC) Mixer: the packet stream from multiple sources may be multiplexed together for transmission Each packet contains a sequence number which is used to detect lost or out-of-sequence packets Time-stamp field indicates the time reference when the packet was created The SSRC indicates from which device the source information has come

12.5 RTP and RTCP RTCP RTP is concerned with the transfer of the individual streams of digitized data associated with a multimedia call The RTCP adds additional system-level functionality to its related RTP QoS reports:the number of lost packets, the level of jitter, and the mean transmission delay The adjoining RTCP sends a message containing the related information to the RTCP in each of these systems at periodic intervals Figure 12.22